[AsteriskBrasil] Problemas com ocupado
Eduardo_Impacto
eduardo em impactovoip.com.br
Quinta Agosto 21 10:42:55 BRT 2008
Alterei a configuração ,mas continua com o mesmo problema
----- Original Message -----
From: Sebastiao Rocha
To: asteriskbrasil em listas.asteriskbrasil.org
Sent: Thursday, August 21, 2008 10:14 AM
Subject: Re: [AsteriskBrasil] Problemas com ocupado
Se usar nat=yes, o canreinvite=no deve ser especificado, não se pode usar os dois juntos.
CanReinvite é usado para que o trafego de voz não passe pelo servidor, ele será feito via RTP entre os ramais, "Peer to Peer" ou IP para IP, como preferir, o NAT não permite que seja estabelecida uma conexão assim.
----- Original Message -----
From: Eduardo_Impacto
To: asteriskbrasil em listas.asteriskbrasil.org
Sent: Thursday, August 21, 2008 9:43 AM
Subject: [AsteriskBrasil] Problemas com ocupado
Bom dia Lista, estou com úm ramal sip só dando ocupado, todas as chamadas que eu realizo elas dão ocupado...abaixo vai o debug, se alguem poder ajudar fico grato
[2454]
type=friend
username=2454
accountcode=2454
regexten=2454
callerid=2401
amaflags=billing
secret=xxxxxxxxxxx
nat=yes
dtmfmode=RFC2833
qualify=yes
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
host=dynamic
context=a2billing
regseconds=0
cancallforward=yes
---
Destroying call '26198a1069cd6c66171b81860ebf9c7a em 201.48.251.15'
Retransmitting #4 (NAT) to 201.22.164.167:5060:
OPTIONS sip:201.22.164.167 SIP/2.0
Via: SIP/2.0/UDP 201.48.251.15:5060;branch=z9hG4bK2232d31a;rport
From: "Unknown" <sip:Unknown em 201.48.251.15>;tag=as1b92be04
To: <sip:201.22.164.167>
Contact: <sip:Unknown em 201.48.251.15>
Call-ID: 390a89934b46b54214b5943e6f33424f em 201.48.251.15
CSeq: 102 OPTIONS
User-Agent: Impacto Voip Pbx
Max-Forwards: 70
Date: Thu, 21 Aug 2008 12:37:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '390a89934b46b54214b5943e6f33424f em 201.48.251.15'
asterisk1*CLI>
<-- SIP read from 201.22.164.167:59317:
INVITE sip:06230911858 em 201.48.251.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;rport
Route: <sip:201.48.251.15:5060;lr>
From: "Claudio" <sip:2454 em 201.48.251.15>;tag=1407274989
To: <sip:06230911858 em 201.48.251.15>
Call-ID: 1762016748-44598-10 em 192.168.0.104
CSeq: 90 INVITE
Contact: <sip:2454 em 192.168.0.104:44598>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXW-4004 V1.1A 1.0.0.67
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 297
v=0
o=2454 8002 8000 IN IP4 192.168.0.104
s=SIP Call
c=IN IP4 192.168.0.104
t=0 0
m=audio 18038 RTP/AVP 18 4 0 8 101
a=sendrecv
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
--- (15 headers 14 lines)---
Using INVITE request as basis request - 1762016748-44598-10 em 192.168.0.104
Sending to 192.168.0.104 : 44598 (NAT)
Reliably Transmitting (NAT) to 201.22.164.167:59317:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;received=201.22.164.167;rport=59317
From: "Claudio" <sip:2454 em 201.48.251.15>;tag=1407274989
To: <sip:06230911858 em 201.48.251.15>;tag=as75d3af08
Call-ID: 1762016748-44598-10 em 192.168.0.104
CSeq: 90 INVITE
User-Agent: Impacto Voip Pbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:06230911858 em 201.48.251.15>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72cfe697"
Content-Length: 0
---
Scheduling destruction of call '1762016748-44598-10 em 192.168.0.104' in 15000 ms
Found user '2454'
asterisk1*CLI>
<-- SIP read from 201.22.164.167:59317:
ACK sip:06230911858 em 201.48.251.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;rport
Route: <sip:201.48.251.15:5060;lr>
From: "Claudio" <sip:2454 em 201.48.251.15>;tag=1407274989
To: <sip:06230911858 em 201.48.251.15>;tag=as75d3af08
Call-ID: 1762016748-44598-10 em 192.168.0.104
CSeq: 90 ACK
Content-Length: 0
--- (8 headers 0 lines)---
asterisk1*CLI>
<-- SIP read from 201.22.164.167:59317:
INVITE sip:06230911858 em 201.48.251.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1021032546;rport
Route: <sip:201.48.251.15:5060;lr>
From: "Claudio" <sip:2454 em 201.48.251.15>;tag=1407274989
To: <sip:06230911858 em 201.48.251.15>
Call-ID: 1762016748-44598-10 em 192.168.0.104
CSeq: 91 INVITE
Contact: <sip:2454 em 192.168.0.104:44598>
Proxy-Authorization: Digest username="2454", realm="asterisk", nonce="72cfe697", uri="sip:06230911858 em 201.48.251.15", response="d223043cc27813ce35691920977491c0", algorithm=MD5
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXW-4004 V1.1A 1.0.0.67
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 297
v=0
o=2454 8002 8000 IN IP4 192.168.0.104
s=SIP Call
c=IN IP4 192.168.0.104
t=0 0
m=audio 18038 RTP/AVP 18 4 0 8 101
a=sendrecv
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
--- (16 headers 14 lines)---
Using INVITE request as basis request - 1762016748-44598-10 em 192.168.0.104
Sending to 192.168.0.104 : 44598 (NAT)
Found user '2454'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.104:18038
Found description format G729
Found description format G723
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 06230911858 in a2billing (domain 201.48.251.15)
Reliably Transmitting (NAT) to 201.22.164.167:59317:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1021032546;received=201.22.164.167;rport=59317
From: "Claudio" <sip:2454 em 201.48.251.15>;tag=1407274989
To: <sip:06230911858 em 201.48.251.15>;tag=as75d3af08
Call-ID: 1762016748-44598-10 em 192.168.0.104
CSeq: 91 INVITE
User-Agent: Impacto Voip Pbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:06230911858 em 201.48.251.15>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 201.22.164.167:59317:
ACK sip:06230911858 em 201.48.251.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1021032546;rport
Route: <sip:201.48.251.15:5060;lr>
From: "Claudio" <sip:2454 em 201.48.251.15>;tag=1407274989
To: <sip:06230911858 em 201.48.251.15>;tag=as75d3af08
Call-ID: 1762016748-44598-10 em 192.168.0.104
CSeq: 91 ACK
Content-Length: 0
Eduardo de Sousa
Departamento Comercial
MSN:atendimento em impactovoip.com.br
Impacto Voip Tecnologia e Teleinformática
www.impactovoip.com.br
Fone: (62) 4053-8840 - 9651-4660
----------------------------------------------------------------------------
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------------------------------------------------------------------------------
_______________________________________________
Compre uma camiseta da AsteriskBrasil.org!
http://www.voipmania.com.br
Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br
_______________________________________________
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AsteriskBrasil em listas.asteriskbrasil.org
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